Monday, 16 June 2014

Bat lures

How to make a bat lure.

Acoustic lures are now sometimes being used for bat surveys whereby ultrasound is used to attract bats to mist nets or harp traps. This can be especially valuable in dense woodland which is difficult to survey, and where bats may be feeding high up in the canopy. The addition of acoustic lures to Natural England Class 19 and 20 licenses also means that more consultants are now able to use this technique.

The broadcast of ultrasound has always been problematic however. Most commercial loudspeakers only work up to the low 20 kHz range, and even then are relatively low-powered. The excellent Ultrasound Advice S55 loudspeaker and amplifier is no longer available which used to be the only commercially available speaker to be able to output at sufficiently high levels over the full ultrasound range used by bats.

There are now a number of commercially available acoustic lure systems, but here I will show how it is possible to make an acoustic lure for bats and generate the sound files necessary based on separate components which are currently available.
The two major limitations are the loudspeaker, and the signal generator.

The speaker

The Pettersson L400 ultrasound loudspeaker is a specialised tweeter housed within a rugged aluminium box and powered by either eight AA batteries or a 12-28 Volt external supply. It can handle a frequency range of between 10-110 kHz at up to 100 dB SPL at 1 m. It has a built-in volume control knob and a female phono connection to accept input signals. It’s available from NHBS at £1400 or direct from Pettersson.

There is also a standard camera tripod mounting bush on one side allowing it to be mounted on a tripod. The frequency response (below) shows it more than capable of outputting signals over the range used for acoustic lures.

The extended frequency response should also allow realistic reproduction of bat echolocation calls, but this would require very high sampling sound cards (which are only now becoming available) or digital to analogue output cards running via data acquisition software such as Labview. There are now some sound cards that operate at 192 kHz which could potentially allow playback of frequencies up to 85 kHz.
The broadcast system

This used to be a severe limitation in the broadcast of ultrasound signals in that you need a sampling rate at least twice that of the highest frequency signal you want to output. The typical industry standard maximum sampling rate was 44.1 kHz, meaning that you could only output signals up to 22 kHz. This would actually work fine for social calls of pipistrelles, but is a bit of a limitation for anything else. However, improvements in technology and demands from consumers for better quality audio has meant that higher sampling rates are now available in small affordable devices, and the new maximum sampling rate often quoted is 96 kHz. This means that it should be possible to output signals at up to 48 kHz, good enough for low frequency bats. The broadcast of full bat echolocation calls is still very specialist and usually requires complex (and expensive) data acquisition cards. However, 48 kHz is more than good enough for an acoustic lure.

One such device is the Roland R-05 recorder (sometimes marketed under the Edirol brand – Edirol being just a brand used by Roland). While this device can sample at up to 96 kHz, so theoretically output up to 48 kHz, the specification shows it limited to 40 kHz, probably due to anti-aliasing filters somewhere along the line. The Zoom devices such as the Zoom H1, H2 and H4 also appear to have 96 kHz sampling rates, so should also be suitable, though I haven’t been able to find any specifications on the actual available bandwidth. If you don’t have one of these devices, you could also use the speaker (line) output from a laptop if it supports the higher sampling rates.
Connecting the two

All you will need to do now is to connect the recorder to the speaker, and for this you will need a 3.5mm stereo jack to phono lead. The jack goes into the recorder, and one of the phono leads attaches to the speaker. For the recorder, it is important to put the lead into the Line-out socket, not the headphones socket. If you recorder does not have a line-out socket, the headphone socket may be configurable as a line-out socket through the device’s menu or by a switch. Have a look at the manual.

As your recorder is a stereo device, it can create two separate tracks for each file, a left and a right. If you create the file on one channel (see below) then you will need to connect the right end of the lead to the phono socket on the loudspeaker. The easy way to remember this is that Red is Right. So if you create your file with the lure sound on the right channel, connect the red lead, if it’s on the left channel, connect the white lead. If it’s on both channels it doesn’t make any difference which lead you use. It’s also possible to create a mono file that doesn’t have two separate channels, so again, connect either lead. I don’t suggest you do this as some modern digital devices have trouble with mono files and don’t know what to do with them. It’s safer to create a stereo file as you know it will work.

Now that you have the hardware set up, all you need to do now is to create the lure sound files.
Creating the lure files

One of the easiest and most versatile pieces of software to generate lure files is Adobe Audition, however, this is a costly piece of software (about £250), so I’ll show how to do it in Audacity, an excellent freeware program available for Windows, Linux and Macs from:

So download and install Audacity, which is quite a small program so installs very quickly.

You should now be looking at something like that below:

The first thing we need to do is to change the project settings to reflect both the correct data type and the correct sampling rate.

Go to Edit>Preferences then select ‘Quality’ and change the settings for the Default Sample rate to 96000 Hz and the Default Sample Format to 16-bit.

The Roland recorder, and many others, are capable of recording and outputting files at 24 bit. However, for our purposes, this just means that the files are bigger and there is a greater risk of them not outputting correctly, so it’s safer to stick to 16 bit.

Then click ‘OK’.

You should now see that the project rate in the bottom left hand corner of the screen says 96000 Hz.

Now we’re ready to generate out lure files.

There is a great deal of debate about what acoustic lure files work best, and it’s still a matter of trial and error, so we’ll create one that has the elements of those which are suggested to work well. Once we can create one, then we can tweak all the components as much as we like.

First, create a new blank track.

Go to Tracks>Add New>Stereo Track

This should create a new blank audio track that we can put out generated signals into. By default, what we generate in one track should also happen in the other track too.

So now we’re looking at two tracks, click the mouse into one of the tracks at the far left hand side.

Then go to Generate>Chirp and in the box that appears, select the waveform as Sine, the start frequency as 35000, the end frequency as 25000, the start amplitude as 0, the end amplitude as 1, the interpolation as logarithmic and the Duration as 00.025 seconds, as shown below.

Then click OK

This will generate a tone that will start quiet at 35 kHz, then sweep down logarithmically to reach maximum intensity at 25 kHz 25 ms later. This is the first half of our tone. We now need to make it sweep back up again.

This time, select the right most side of the signal you have just generated, but now go to Generate>Chirp, but do the opposite, start at 25000 and intensity 1, and end at 35000 intensity 0, with the duration still set to 0.025 seconds.

If you now go to View>Fit In Window, you should see the signal looking something like this:

So now we have a signal that starts at 35 kHz, sweeps down to 25 kHz, then back up to 35 kHz in 50 ms.

The sonogram, (in Batsound) looks like this:

Notice that there is a little frequency  ‘blip’ at the bottom of the sweep, this is caused by the joining of the two signals that are not quite in phase. This is a little messy, which is why it’s easier to generate these files in Adobe Audition where we can generate these frequency modulations more easily, but we can live with this.

Now we have one sweep cycle, we can just copy and paste the generated file as often as we like to build up a sequence.

Click the mouse onto the left hand of the waveform and either drag it to the right hand side, or click over the left hand side of the waveform and hold down the Shift key and use the right hand arrow button to select the whole waveform. This is better as the selection stops when you reach the right hand side.

The select Edit>Copy and click or use the cursor keys to select the right hand end of the waveform and lick Edit>Paste or use Control+V and you’ve added a copy of the waveform, so now you have two up and down cycles. If you can’t see it all, go to View>Fit In Window.

Now go to the end of the next waveform and press Control+V again, and you get another one, go to the end of that one and press Control+V and so on. I’d suggest perhaps ten cycles, so the total lasts now 500 ms. In Batsound it now looks like this.

Now we could leave it at that, and it will work, but it’s nice to have a bit of a lead in and lead out time, so we’ll put some silences at the beginning and end, I suggest  500 ms will do.

Select the very start of your sequence, and then go to Generate>Silence. In the box put in 00.500s

Click OK, then go to View>Fit in Window.

Do the same at the end of the track to add a 500 ms silence at the end.

And that’s it!

The final file looks like this:

You can now save your file as a .wav file onto the memory card of your recorder go to File>Export  and choose WAV file format, paying attention to any specific file format rules that might apply to your recorder. The Roland R-05 should be able to play files with any name.

After exporting your .wav file, you can save your project file, it is saved as an .aup file which audacity can then edit again.

If you have a repeat play function on your recorder, then simply play the file but set it to repeat over and over. If you don’t have a repeat play option, then simply copy and then paste the whole of your sequence at the end of the preceding part to double it up, and then copy and paste again to double that up and so on until you have a sequence the length you need.

Once you have mastered the general principles you can create files of almost any structure by combining sweeps, tones, silences, noise and so on.

If you want to make more complex amplitude modulations (i.e. change the loudness up and down), then a good way is to use the Envelope Tool, which allows you to select different parts of the waveform and then drag their amplitude up and down as shown below.

Audacity is a very sophisticated program, and there are better ways of achieving the same end but this requires the use of scripting to combine two or more waveforms to achieve both the frequency modulation and the amplitude modulation.

If the clicks bother you at the bottom of the waveforms then you can use Audacity to high pass the signal above 20 kHz (go to Effect>High Pass Filter then set the slider to 20000 Hz).  It may be that the bats like the clicks. The signals that they seem attracted to bear no relation to their normal social calls, perhaps they’re just very inquisitive animals?
Using real social calls

One final note is that if you have recordings of real social calls, you might want to use those instead. However, you will need to convert the sample rate to that of the output device. In principle, this is can be easily done in Audacity, but there are a number of issues to consider.

 If you have files from a D500x, D1000x or other device that streams the sound files to a memory card, then the file header has the real sample rate embedded within it, such as 500 kHz. In this case if you simply open the file and go to the Tracks>Resample option and type in the new sample rate, Audacity will resample the track down to the new sample rate which should match that of the recorder i.e. 96 kHz to maintain both the duration and the frequencies of the original signal. However, you are in effect aliasing the signal by resampling at too low a sample rate to preserve all the information. This means that parts of the signal higher than 48 kHz will be aliased which may create some unwanted lower frequency elements. Anything below 48 kHz should remain unaffected. It does seem though that Audacity will automatically low-pass the signal before resampling, meaning that aliasing should not be a problem, though obviously anything above 40 kHz will be lost.
The situation is a little more complicated for time-expanded signals which have been recorded on an external recorder. Say a 2 second sequence has been recorded and output as 10x time expansion so that it lasts 20 seconds and recorded onto an external device at a sampling rate of 44.1 kHz. If we resampled this signal up to 96 kHz, we would still have a 20 second long track where all the frequencies were 10x lower than we needed, we would just have a signal with a higher sampling rate.

First we need to change the sample rate in the file header to make the sequence 2 seconds long again, and so with the drop down box from the ‘Audio Track’ option to the right of the track, select ‘Set Rate’ and choose a sample rate 10x that of the original file – if it was 44.1 kHz, select  Other’ and type in 441,000 Hz (note that Audacity works in Hz rather than kHz so always multiply any kHz frequencies by 1,000).

Now we have a file of the right length (2 seconds) and with the right frequencies, we have to choose the Tracks>Resample option and set it to 96 kHz, to allow out recorder to play it.
It should be pointed out that resampling can sometimes introduce some nasty artefacts into recordings, so always look (and listen) to the recordings afterwards to make sure they are faithful to the original.
I should add of course that ultrasound is very directional, and will be beamed quite tightly from the loudspeaker, higher frequencies being more directional. So a low frequency (15-20 kHz) signal will be broadcast quite widely, while a 40 kHz signal will be pretty directional. The Sussex Autobat has the famous 'twiddler' that broadcasts that signal around (and also adds some amplitude and frequency modulations that appear to make the signal more attractive), so using the L400 it would be best to change the direction of broadcast frequently during a session.


  1. For anyone keen on using Audacity and finding Adobe's subscription programme a little expensive to get Audition, there are some scripts here which will allow you to generate FM signals in Audacity, something that Audition was better at. Audacity is free and very flexible though has a bit of a steep learning curve.

  2. It was another joy to see your post. It is such an important topic and ignored by so many, even professionals. I thank you to help making people more aware of possible issues. Great stuff as usual...
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